/*
* Butterworth Filter for HP or LP
* out(n) = a1 * in + a2 * in(n-1) + a3 * in(n-2) - b1*out(n-1) - b2*out(n-2)
* args:
* FILT - pointer to fx_filt_data_t
* Buff - sampe buffer
* NumSamples - samples in buffer
* channels - number of audio channels
*/
static void Butt(fx_filt_data_t *FILT,
sample_t *Buff,
int NumSamples,
int channels)
{
int index = 0;
for (index = 0; index < NumSamples; index = index + channels)
{
sample_t out = FILT->a1 * Buff[index] + FILT->a2 * FILT->buff_in1[0] +
FILT->a3 * FILT->buff_in1[1] - FILT->b1 * FILT->buff_out1[0] -
FILT->b2 * FILT->buff_out1[1];
FILT->buff_in1[1] = FILT->buff_in1[0]; //in(n-2) = in(n-1)
FILT->buff_in1[0] = Buff[index]; // in(n-1) = in
FILT->buff_out1[1] = FILT->buff_out1[0]; //out(n-2) = out(n-1)
FILT->buff_out1[0] = out; //out(n-1) = out
Buff[index] = clip_float(out);
/*process second channel*/
if(channels > 1)
{
out = FILT->a1 * Buff[index+1] + FILT->a2 * FILT->buff_in2[0] +
FILT->a3 * FILT->buff_in2[1] - FILT->b1 * FILT->buff_out2[0] -
FILT->b2 * FILT->buff_out2[1];
FILT->buff_in2[1] = FILT->buff_in2[0]; //in(n-2) = in(n-1)
FILT->buff_in2[0] = Buff[index+1]; // in(n-1) = in
FILT->buff_out2[1] = FILT->buff_out2[0]; //out(n-2) = out(n-1)
FILT->buff_out2[0] = out; //out(n-1) = out
Buff[index+1] = clip_float(out);
}
}
}
/*
* Echo effect
* args:
* audio_ctx - audio context
* data - audio buffer to be processed
* delay_ms - echo delay in ms (e.g: 300)
* decay - feedback gain (<1) (e.g: 0.5)
*
* asserts:
* audio_ctx is not null
*
* returns: none
*/
static void audio_fx_echo(audio_context_t *audio_ctx,
sample_t *data,
int delay_ms,
float decay)
{
/*assertions*/
assert(audio_ctx != NULL);
int samp = 0;
if(aud_fx->ECHO == NULL)
{
aud_fx->ECHO = calloc(1, sizeof(fx_delay_data_t));
if(aud_fx->ECHO == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (audio_fx_echo): %s\n", strerror(errno));
exit(-1);
}
aud_fx->ECHO->buff_size = (int) delay_ms * audio_ctx->samprate * 0.001;
aud_fx->ECHO->delayBuff1 = calloc(aud_fx->ECHO->buff_size, sizeof(sample_t));
if(aud_fx->ECHO->delayBuff1 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (audio_fx_echo): %s\n", strerror(errno));
exit(-1);
}
aud_fx->ECHO->delayBuff2 = NULL;
if(audio_ctx->channels > 1)
{
aud_fx->ECHO->delayBuff2 = calloc(aud_fx->ECHO->buff_size, sizeof(sample_t));
if(aud_fx->ECHO->delayBuff2 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (audio_fx_echo): %s\n", strerror(errno));
exit(-1);
}
}
}
for(samp = 0; samp < audio_ctx->capture_buff_size; samp = samp + audio_ctx->channels)
{
sample_t out = (0.7 * data[samp]) +
(0.3 * aud_fx->ECHO->delayBuff1[aud_fx->ECHO->delayIndex]);
aud_fx->ECHO->delayBuff1[aud_fx->ECHO->delayIndex] = data[samp] +
(aud_fx->ECHO->delayBuff1[aud_fx->ECHO->delayIndex] * decay);
data[samp] = clip_float(out);
/*if stereo process second channel in separate*/
if (audio_ctx->channels > 1)
{
out = (0.7 * data[samp+1]) +
(0.3 * aud_fx->ECHO->delayBuff2[aud_fx->ECHO->delayIndex]);
aud_fx->ECHO->delayBuff2[aud_fx->ECHO->delayIndex] = data[samp] +
(aud_fx->ECHO->delayBuff2[aud_fx->ECHO->delayIndex] * decay);
data[samp+1] = clip_float(out);
}
if(++(aud_fx->ECHO->delayIndex) >= aud_fx->ECHO->buff_size) aud_fx->ECHO->delayIndex=0;
}
}
/*
* WahWah effect
* !!!!!!!!!!!!! IMPORTANT!!!!!!!!! :
* depth and freqofs should be from 0(min) to 1(max) !
* res should be greater than 0 !
* args:
* audio_ctx - audio context
* data - audio buffer to be processed
* freq - LFO frequency (1.5)
* startphase - LFO startphase in RADIANS - usefull for stereo WahWah (0)
* depth - Wah depth (0.7)
* freqofs - Wah frequency offset (0.3)
* res - Resonance (2.5)
*
* asserts:
* audio_ctx is not null
*
* returns: none
*/
static void audio_fx_wahwah (audio_context_t *audio_ctx,
sample_t *data,
float freq,
float startphase,
float depth,
float freqofs,
float res)
{
/*assertions*/
assert(audio_ctx != NULL);
float frequency, omega, sn, cs, alpha;
if(aud_fx->wahData == NULL)
{
aud_fx->wahData = calloc(1, sizeof(fx_wah_data_t));
if(aud_fx->wahData == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (audio_fx_wahwah): %s\n", strerror(errno));
exit(-1);
}
aud_fx->wahData->lfoskip = freq * 2 * M_PI / audio_ctx->samprate;
aud_fx->wahData->phase = startphase;
/*if right channel set: phase += (float)M_PI;*/
}
int samp = 0;
for(samp = 0; samp < audio_ctx->capture_buff_size; samp++)
{
float in = data[samp];
if ((aud_fx->wahData->skipcount++) % lfoskipsamples == 0)
{
frequency = (1 + cos(aud_fx->wahData->skipcount * aud_fx->wahData->lfoskip + aud_fx->wahData->phase)) * 0.5;
frequency = frequency * depth * (1 - freqofs) + freqofs;
frequency = exp((frequency - 1) * 6);
omega = M_PI * frequency;
sn = sin(omega);
cs = cos(omega);
alpha = sn / (2 * res);
aud_fx->wahData->b0 = (1 - cs) * 0.5;
aud_fx->wahData->b1 = 1 - cs;
aud_fx->wahData->b2 = (1 - cs) * 0.5;
aud_fx->wahData->a0 = 1 + alpha;
aud_fx->wahData->a1 = -2 * cs;
aud_fx->wahData->a2 = 1 - alpha;
}
float out = (aud_fx->wahData->b0 * in + aud_fx->wahData->b1 * aud_fx->wahData->xn1 +
aud_fx->wahData->b2 * aud_fx->wahData->xn2 - aud_fx->wahData->a1 * aud_fx->wahData->yn1 -
aud_fx->wahData->a2 * aud_fx->wahData->yn2) / aud_fx->wahData->a0;
aud_fx->wahData->xn2 = aud_fx->wahData->xn1;
aud_fx->wahData->xn1 = in;
aud_fx->wahData->yn2 = aud_fx->wahData->yn1;
aud_fx->wahData->yn1 = out;
data[samp] = clip_float(out);
}
}
void GenEVNODD(DspParamData& fblk)
{
fblk._mods._evaluator = [=](FPARAM& cec) -> float
{
float kt = fblk._keyTrack*cec._keyOff;
float vt = lerp(-fblk._velTrack,0.0f,cec._unitVel);
float x = (cec._coarse)+cec._C1()+cec._C2()+kt+vt;
//printf( "vt<%f> kt<%f> x<%f>\n", vt, kt, x );
return clip_float(x,-10,10);
};
}
void TWOPARAM_SHAPER::compute(DspBuffer& dspbuf) //final
{
float pad = _dbd._inputPad;
int inumframes = dspbuf._numframes;
float* ubuf = dspbuf.channel(0);
float evn = _param[0].eval();
float odd = _param[1].eval();
//evn = -22;//(0.5f+sinf(ph1+pi)*0.5f)*-60.0f;
//odd = (sinf(ph1)*30.f)-30.0f;
ph1 += 0.0003f;
_fval[0]=evn;
_fval[1]=odd;
//printf( "_dbd._inputPad<%f>\n", _dbd._inputPad );
if(1) for( int i=0; i<inumframes; i++ )
{
float u = ubuf[i]*pad;
float usq = u*u;
float le = usq*decibel_to_linear_amp_ratio(evn);
float lo = u*decibel_to_linear_amp_ratio(odd);
//float e = (2.0f*powf(le,2.0f))-1.0f;
//float e = ((2.0f*powf(le,2.0f))-1.0f)*1000.0f;//decibel_to_linear_amp_ratio(-evn);
float index = clip_float(le*6,-12,12); //clip_float(powf(le,4),-12.0f,12.0f);
// index *= index;
float e = sinf(index*pi2); ///adj;
index = clip_float(lo*6,-12.0f,12.0f);
float o = sinf(index*pi2); ///adj;
float r = (e+o)*0.5f;
//r = wrap(r,-30);
ubuf[i] = r;
}
}
void GenPOS(DspParamData& fblk)
{
fblk._mods._evaluator = [=](FPARAM& cec) -> float
{
cec._kval = fblk._keyTrack*cec._keyOff;
cec._vval = fblk._velTrack*cec._unitVel;
cec._s1val = cec._C1();
cec._s2val = cec._C2();
float x = (cec._coarse)
+ cec._s1val
+ cec._s2val
+ cec._kval
+ cec._vval;
return clip_float(x,-100,100);
};
}
controller_t layer::getSRC2(const BlockModulationData& mods)
{
auto src2 = this->getController(mods._src2);
auto depthcon = this->getController(mods._src2DepthCtrl);
float mindepth = mods._src2MinDepth;
float maxdepth = mods._src2MaxDepth;
auto it = [=]()->float
{
float dc = clip_float(depthcon(),0,1);
float depth = lerp(mindepth,maxdepth,dc);
float out = src2()*depth;
return out;
};
return it;
}
/* Non-linear amplifier with soft distortion curve.
* args:
* input - sample input
*
* asserts:
* none
*
* returns: processed sample
*/
static sample_t CubicAmplifier( sample_t input )
{
sample_t out;
float temp;
if( input < 0 ) /*silence*/
{
temp = input + 1.0f;
out = (temp * temp * temp) - 1.0f;
}
else
{
temp = input - 1.0f;
out = (temp * temp * temp) + 1.0f;
}
return clip_float(out);
}
void layer::compute(outputBuffer& obuf)
{
_HAF._items.clear();
///////////////////////
if( nullptr == _layerData )
{
printf( "gotnull ld layer<%p>\n", this );
return;
}
if( nullptr == _alg )
return;
//printf( "layer<%p> compute\n", this );
int inumframes = obuf._numframes;
float* outl = obuf._leftBuffer;
float* outr = obuf._rightBuffer;
float dt = float(inumframes)/_syn._sampleRate;
//////////////////////////////////////
// update controllers
//////////////////////////////////////
for( int i=0; i<kmaxctrlblocks; i++ )
{
if( _ctrlBlock[i] )
_ctrlBlock[i]->compute(inumframes);
}
///////////////////////////////////
// extract AMPENV
///////////////////////////////////
// TODO - get rid og hard coded location in CB0
auto CB0 = this->_ctrlBlock[0];
if( CB0 )
{
if( auto as_env = dynamic_cast<RateLevelEnvInst*>(CB0->_cinst[0] ) ) // ampenv ?
{
for( int i=0; i<inumframes; i++ )
{
float e0 = as_env->_USERAMPENV[i];
_USERAMPENV[i] = e0;
}
}
}
///////////////////////////////////////////////
// HUD AFRAME
///////////////////////////////////////////////
int envcount = 0;
int asrcount = 0;
int lfocount = 0;
int funcount = 0;
for( int icb=0; icb<kmaxctrlblocks; icb++ )
{ auto cb = _ctrlBlock[icb];
if( cb )
{ for( int ic=0; ic<kmaxctrlperblock; ic++)
{ auto cinst = cb->_cinst[ic];
if( auto env = dynamic_cast<RateLevelEnvInst*>(cinst) )
{ envframe envf;
envf._index = envcount++;
envf._value = env->_curval;
envf._data = env->_data;
envf._curseg = env->_curseg;
if( env->_data && env->_data->_ampenv )
envf._curseg = _useNatEnv ? _HAF_nenvseg : env->_curseg;
_HAF._items.push_back(envf);
}
else if( auto asr = dynamic_cast<AsrInst*>(cinst) )
{ asrframe asrf;
asrf._index = asrcount++;
asrf._value = asr->_curval;
asrf._curseg = asr->_curseg;
asrf._data = asr->_data;
_HAF._items.push_back(asrf);
//printf( "add asr item\n");
}
else if( auto lfo = dynamic_cast<LfoInst*>(cinst) )
{ lfoframe lfof;
lfof._index = lfocount++;
lfof._value = lfo->_curval;
lfof._currate = lfo->_currate;
lfof._data = lfo->_data;
_HAF._items.push_back(lfof);
}
else if( auto fun = dynamic_cast<FunInst*>(cinst) )
{ funframe funfr;
funfr._index = funcount++;
funfr._data = fun->_data;
funfr._value = fun->_curval;
_HAF._items.push_back(funfr);
}
}
}
}
///////////////////////////////////////////////
auto PCHBLK = _layerData->_dspBlocks[0];
if( PCHBLK )
{
const int kNOTEC4 = 60;
const auto& PCH = PCHBLK->_paramd[0];
const auto& KMP = _layerData->_kmpBlock;
int timbreshift = KMP._timbreShift; // 0
int kmtrans = KMP._transpose/*+timbreshift*/; // -20
int kmkeytrack = KMP._keyTrack; // 100
int kmpivot = (kNOTEC4+kmtrans); // 48-20 = 28
int kmdeltakey = (_curnote+kmtrans-kmpivot); // 48-28 = 28
int kmdeltacents = kmdeltakey*kmkeytrack; // 8*0 = 0
int kmfinalcents = (kmpivot*100)+kmdeltacents; // 4800
int pchtrans = PCH._coarse;//-timbreshift; // 8
int pchkeytrack = PCH._keyTrack; // 0
int pchpivot = (kNOTEC4+pchtrans); // 48-0 = 48
int pchdeltakey = (_curnote+pchtrans-pchpivot); // 48-48=0 //possible minus kmorigin?
int pchdeltacents = pchdeltakey*pchkeytrack;//0*0=0
int pchfinalcents = (pchtrans*100)+pchdeltacents;//0*100+0=0
int kmcents = kmfinalcents;//+region->_tuning;
//printf( "_curCentsOSC<%d>\n", int(_curCentsOSC) );
auto dsp0 = _alg->_block[0];
if( dsp0 )
{
float centoff = dsp0->_param[0].eval();
_curPitchOffsetInCents = int(centoff);//kmcents+pchfinalcents;
}
_curCentsOSC = kmcents+pchfinalcents + _curPitchOffsetInCents;
}
else
{
_curCentsOSC = _curnote*100;
}
_curCentsOSC = clip_float( _curCentsOSC, -0,12700 );
//printf( "pchc1<%f> pchc2<%f> poic<%f> currat<%f>\n", _pchc1, _pchc2, _curPitchOffsetInCents, currat );
////////////////////////////////////////
//printf( "doBlockStereo<%d>\n", int(doBlockStereo) );
////////////////////////////////////////
if( true )
{
bool bypassDSP = _syn._bypassDSP;
DspBlock* lastblock = _alg->lastBlock();
bool doBlockStereo = bypassDSP
? false
: lastblock ? (lastblock->numOutputs()==2)
: false;
float synsr = _syn._sampleRate;
outputBuffer laybuf;
_layerObuf.resize(inumframes);
float* lyroutl = _layerObuf._leftBuffer;
float* lyroutr = _layerObuf._rightBuffer;
///////////////////////////////////
// sample osc
///////////////////////////////////
bool do_noise = _doNoise;
bool do_sine = false;
bool do_input = false;
switch(_syn._genmode )
{
case 1: // force sine
do_sine = true;
do_noise = false;
break;
case 2: // force noise
do_sine = false;
do_noise = true;
break;
case 3: // input
do_sine = false;
do_noise = false;
do_input = true;
break;
}
if( do_noise )
{
for( int i=0; i<inumframes; i++ )
{
float o = ((rand()&0xffff)/32768.0f)-1.0f;
lyroutl[i] = o;
lyroutr[i] = 0.0f;
}
}
else if( do_input )
{
auto ibuf = _syn._ibuf._leftBuffer;
for( int i=0; i<inumframes; i++ )
{
float o = ibuf[i]*8.0f;
lyroutl[i] = o;
lyroutr[i] = 0.0f;
}
}
else if( do_sine )
{
float F = midi_note_to_frequency(float(_curCentsOSC)*0.01);
float phaseinc = pi2*F/synsr;
for( int i=0; i<inumframes; i++ )
{
float o = sinf(_sinrepPH)*0.5;
_sinrepPH += phaseinc;
lyroutl[i] = o;
lyroutr[i] = 0.0f;
}
}
else // clear
for( int i=0; i<inumframes; i++ )
{
lyroutl[i] = 0.0f;
lyroutr[i] = 0.0f;
}
///////////////////////////////////
// DSP F1-F3
///////////////////////////////////
if( false==bypassDSP )
_alg->compute(_layerObuf);
///////////////////////////////////
// amp / out
///////////////////////////////////
if( doBlockStereo )
{
for( int i=0; i<inumframes; i++ )
{
float tgain = _layerGain*_masterGain;
outl[i] += lyroutl[i]*tgain;
outr[i] += lyroutr[i]*tgain;
}
}
else if( bypassDSP )
{
for( int i=0; i<inumframes; i++ )
{
float tgain = _layerGain*_masterGain;
float inp = lyroutl[i];
outl[i] += inp*tgain*0.5f;
outr[i] += inp*tgain*0.5f;
}
}
else
{
for( int i=0; i<inumframes; i++ )
{
float tgain = _layerGain*_masterGain;
outl[i] += lyroutl[i]*tgain;
outr[i] += lyroutl[i]*tgain;
}
}
/////////////////
// oscope
/////////////////
if( this == _syn._hudLayer )
{
_HAF._oscopebuffer.resize(inumframes);
if( doBlockStereo )
{
for( int i=0; i<inumframes; i++ )
{
float l = _layerObuf._leftBuffer[i];
float r = _layerObuf._rightBuffer[i];
//tailb[i] = l;//doBlockStereo ? l+r : l;
_HAF._oscopebuffer[i]=(l+r)*0.5f;
}
}
else
{
for( int i=0; i<inumframes; i++ )
{
float l = _layerObuf._leftBuffer[i];
float r = _layerObuf._rightBuffer[i];
//tailb[i] = l;//doBlockStereo ? l+r : l;
_HAF._oscopebuffer[i]=l;
}
}
_syn._hudbuf.push(_HAF);
}
}
_layerTime += dt;
}
/*
* four paralell Comb filters for reverb
* args:
* audio_ctx - audio context
* data - audio buffer to be processed
* delay1_ms - delay for filter 1
* delay2_ms - delay for filter 2
* delay3_ms - delay for filter 3
* delay4_ms - delay for filter 4
* gain1 - feed gain for filter 1
* gain2 - feed gain for filter 2
* gain3 - feed gain for filter 3
* gain4 - feed gain for filter 4
* in_gain - input line gain
*
* asserts:
* none
*
* returns: none
*/
static void CombFilter4 (audio_context_t *audio_ctx,
sample_t *data,
int delay1_ms,
int delay2_ms,
int delay3_ms,
int delay4_ms,
float gain1,
float gain2,
float gain3,
float gain4,
float in_gain)
{
int samp=0;
/*buff_size in samples*/
if (aud_fx->COMB4 == NULL)
{
aud_fx->COMB4 = calloc(1, sizeof(fx_comb4_data_t));
if(aud_fx->COMB4 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
/*buff_size in samples*/
aud_fx->COMB4->buff_size1 = (int) delay1_ms * (audio_ctx->samprate * 0.001);
aud_fx->COMB4->buff_size2 = (int) delay2_ms * (audio_ctx->samprate * 0.001);
aud_fx->COMB4->buff_size3 = (int) delay3_ms * (audio_ctx->samprate * 0.001);
aud_fx->COMB4->buff_size4 = (int) delay4_ms * (audio_ctx->samprate * 0.001);
aud_fx->COMB4->CombBuff10 = calloc(aud_fx->COMB4->buff_size1, sizeof(sample_t));
if(aud_fx->COMB4->CombBuff10 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
aud_fx->COMB4->CombBuff20 = calloc(aud_fx->COMB4->buff_size2, sizeof(sample_t));
if(aud_fx->COMB4->CombBuff20 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
aud_fx->COMB4->CombBuff30 = calloc(aud_fx->COMB4->buff_size3, sizeof(sample_t));
if(aud_fx->COMB4->CombBuff30 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
aud_fx->COMB4->CombBuff40 = calloc(aud_fx->COMB4->buff_size4, sizeof(sample_t));
if(aud_fx->COMB4->CombBuff40 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
aud_fx->COMB4->CombBuff11 = NULL;
aud_fx->COMB4->CombBuff21 = NULL;
aud_fx->COMB4->CombBuff31 = NULL;
aud_fx->COMB4->CombBuff41 = NULL;
if(audio_ctx->channels > 1)
{
aud_fx->COMB4->CombBuff11 = calloc(aud_fx->COMB4->buff_size1, sizeof(sample_t));
if(aud_fx->COMB4->CombBuff11 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
aud_fx->COMB4->CombBuff21 = calloc(aud_fx->COMB4->buff_size2, sizeof(sample_t));
if(aud_fx->COMB4->CombBuff21 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
aud_fx->COMB4->CombBuff31 = calloc(aud_fx->COMB4->buff_size3, sizeof(sample_t));
if(aud_fx->COMB4->CombBuff31 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
aud_fx->COMB4->CombBuff41 = calloc(aud_fx->COMB4->buff_size4, sizeof(sample_t));
if(aud_fx->COMB4->CombBuff41 == NULL)
{
fprintf(stderr,"AUDIO: FATAL memory allocation failure (CombFilter4): %s\n", strerror(errno));
exit(-1);
}
}
}
for(samp = 0; samp < audio_ctx->capture_buff_size; samp = samp + audio_ctx->channels)
{
sample_t out1 = in_gain * data[samp] +
gain1 * aud_fx->COMB4->CombBuff10[aud_fx->COMB4->CombIndex1];
sample_t out2 = in_gain * data[samp] +
gain2 * aud_fx->COMB4->CombBuff20[aud_fx->COMB4->CombIndex2];
sample_t out3 = in_gain * data[samp] +
gain3 * aud_fx->COMB4->CombBuff30[aud_fx->COMB4->CombIndex3];
sample_t out4 = in_gain * data[samp] +
gain4 * aud_fx->COMB4->CombBuff40[aud_fx->COMB4->CombIndex4];
aud_fx->COMB4->CombBuff10[aud_fx->COMB4->CombIndex1] = data[samp] +
gain1 * aud_fx->COMB4->CombBuff10[aud_fx->COMB4->CombIndex1];
aud_fx->COMB4->CombBuff20[aud_fx->COMB4->CombIndex2] = data[samp] +
gain2 * aud_fx->COMB4->CombBuff20[aud_fx->COMB4->CombIndex2];
aud_fx->COMB4->CombBuff30[aud_fx->COMB4->CombIndex3] = data[samp] +
gain3 * aud_fx->COMB4->CombBuff30[aud_fx->COMB4->CombIndex3];
aud_fx->COMB4->CombBuff40[aud_fx->COMB4->CombIndex4] = data[samp] +
gain4 * aud_fx->COMB4->CombBuff40[aud_fx->COMB4->CombIndex4];
data[samp] = clip_float(out1 + out2 + out3 + out4);
/*if stereo process second channel */
if(audio_ctx->channels > 1)
{
out1 = in_gain * data[samp+1] +
gain1 * aud_fx->COMB4->CombBuff11[aud_fx->COMB4->CombIndex1];
out2 = in_gain * data[samp+1] +
gain2 * aud_fx->COMB4->CombBuff21[aud_fx->COMB4->CombIndex2];
out3 = in_gain * data[samp+1] +
gain3 * aud_fx->COMB4->CombBuff31[aud_fx->COMB4->CombIndex3];
out4 = in_gain * data[samp+1] +
gain4 * aud_fx->COMB4->CombBuff41[aud_fx->COMB4->CombIndex4];
aud_fx->COMB4->CombBuff11[aud_fx->COMB4->CombIndex1] = data[samp+1] +
gain1 * aud_fx->COMB4->CombBuff11[aud_fx->COMB4->CombIndex1];
aud_fx->COMB4->CombBuff21[aud_fx->COMB4->CombIndex2] = data[samp+1] +
gain2 * aud_fx->COMB4->CombBuff21[aud_fx->COMB4->CombIndex2];
aud_fx->COMB4->CombBuff31[aud_fx->COMB4->CombIndex3] = data[samp+1] +
gain3 * aud_fx->COMB4->CombBuff31[aud_fx->COMB4->CombIndex3];
aud_fx->COMB4->CombBuff41[aud_fx->COMB4->CombIndex4] = data[samp+1] +
gain4 * aud_fx->COMB4->CombBuff41[aud_fx->COMB4->CombIndex4];
data[samp+1] = clip_float(out1 + out2 + out3 + out4);
}
if(++(aud_fx->COMB4->CombIndex1) >= aud_fx->COMB4->buff_size1) aud_fx->COMB4->CombIndex1=0;
if(++(aud_fx->COMB4->CombIndex2) >= aud_fx->COMB4->buff_size2) aud_fx->COMB4->CombIndex2=0;
if(++(aud_fx->COMB4->CombIndex3) >= aud_fx->COMB4->buff_size3) aud_fx->COMB4->CombIndex3=0;
if(++(aud_fx->COMB4->CombIndex4) >= aud_fx->COMB4->buff_size4) aud_fx->COMB4->CombIndex4=0;
}
}
